QRQ CW audio tones over Gstreamer's AUDIO PIPELINES - 16 bit audio VS 24 bit audio - does 24 bit audio make any difference for QRQcw COPY by ear ?

[Chuck aa0hw]

brief test creating 2 Gstreamer AUDIO PIPELINE LOOPBACKS from a Linux Laptop TO a RASPBERRY PI 2b - over the LAN - going through routers and switches...(both using Jack Audio Connection Kit)
one PIPELINE using rtpL16***
and the other PIPELINE using rtpL24***

to compare Gstreamer's rtpL16pay and rtpL24pay RTP AUDIO PAYLOADER ENCODERs , for their inherent NOISE LEVELS before any audio is transmitted into the PIPELINE and then to see how well MORSE CODE AUDIO TONES are preserved(in how they look and sound compared to the original), after going through each of the Gstreamer Audio PIPELINES

RESULTS: there are some wide spectrum noise levels residing on the rtpL16pay PIPES that were not observed on the rtpL24pay PIPES, while viewing the SPECTRUM ANALYZER ; otherwise, the sound of the MORSE CODE AUDIO TONES were pretty close in audio fidelity and spectral bandwidth(above -60db) and shape between rtpL16 & rtpL24

GSTREAMER COMMANDS USED:

RASPBERRY PI:
********************************************************************
gst-launch-1.0 -v jackaudiosrc client-name=TRANSMIT_rtpL24pay provide-clock=true do-timestamp=true ! audio/x-raw, channels=1,depth=24,width=24,rate=48000,payload=96 ! audioconvert ! rtpL24pay ! udpsink host=192.168.1.101 port=5001

gst-launch-1.0 udpsrc port=5002 ! "application/x-rtp,media=(string)audio, clock-rate=(int)48000, channels=1, payload=(int)96" ! rtpjitterbuffer latency=10 ! rtpL24depay ! audioconvert ! jackaudiosink client-name=RECEIVE_rtpL24depay buffer-time=30000

gst-launch-1.0 -v jackaudiosrc client-name=TRANSMIT_rtpL16pay provide-clock=true do-timestamp=true ! audio/x-raw, channels=1,depth=16,width=16,rate=48000,payload=96 ! audioconvert ! rtpL16pay ! udpsink host=192.168.1.101 port=5003

gst-launch-1.0 udpsrc port=5004 ! "application/x-rtp,media=(string)audio, clock-rate=(int)48000, channels=1, payload=(int)96" ! rtpjitterbuffer latency=10 ! rtpL16depay ! audioconvert ! jackaudiosink client-name=RECEIVE_rtpL16depay buffer-time=30000

LINUX LAPTOP:
********************************************************************
gst-launch-1.0 -v jackaudiosrc client-name=TRANSMIT_rtpL24pay provide-clock=true do-timestamp=true ! audio/x-raw, channels=1,depth=24,width=24,rate=48000,payload=96 ! audioconvert ! rtpL24pay ! udpsink host=192.168.1.130 port=5002

gst-launch-1.0 udpsrc port=5001 ! "application/x-rtp,media=(string)audio, clock-rate=(int)48000, channels=1, payload=(int)96" ! rtpjitterbuffer latency=10 ! rtpL24depay ! audioconvert ! jackaudiosink client-name=RECEIVE_rtpL24depay buffer-time=30000

gst-launch-1.0 -v jackaudiosrc client-name=TRANSMIT_rtpL16pay provide-clock=true do-timestamp=true ! audio/x-raw, channels=1,depth=16,width=16,rate=48000,payload=96 ! audioconvert ! rtpL16pay ! udpsink host=192.168.1.130 port=5004

gst-launch-1.0 udpsrc port=5003 ! "application/x-rtp,media=(string)audio, clock-rate=(int)48000, channels=1, payload=(int)96" ! rtpjitterbuffer latency=10 ! rtpL16depay ! audioconvert ! jackaudiosink client-name=RECEIVE_rtpL16depay buffer-time=30000